[Asterisk-Users] FW: Cisco 7960, SIP, NAT, Voicemal

Adams, Gavin gadams at promisant.com
Mon Aug 4 15:12:08 MST 2003


-----Original Message-----
From: Adams, Gavin 
Sent: Monday, August 04, 2003 6:10 PM
To: 'asterisk-users at lists.digium.com'
Subject: Cisco 7960, SIP, NAT, Voicemal

Hey all,

I've got a couple 79xx phones working peer-to-peer and am now trying to
work on the voice mail.

In extensions.conf:

[ATL]
exten => 4001,1,Dial(SIP/gadams)|10
exten => 4001,2,Voicemail,u4001
exten => 4001,102,Voicemail,b4001

and the corresponding sip.conf:

[gadams]
type=friend
username=gadams
secret=******
context=ATL
host=dynamic
canreinvite=no
nat=yes
mailbox=4001

When this phone is dialed, it doesn't roll over to VM after 10 seconds
but continues to ring. If the calling party hangs up, the phone
continues to ring.

However, as a test I changed the |10 to a |10t. At that point dialing
4001 did indeed roll over to voicemail, but it happened immediately.
Also, I'm getting the following message during the dial:

WARNING[1133735216]: File chan_sip.c, Line 417 (retrans_pkt): Maximum
retries exceeded on call 71817d20169869e82ee3d897699393e3 at 63.111.7.161
for seqno 102 (Request)

Which is tied to the call in question. Any clues?

--- Gavin




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